WO2010012155A1

WO2010012155A1 - Method for adaptively adjusting receiving rate,buffering and playing of mobile multimedia broadcast terminal

Info

Publication number: WO2010012155A1
Application number: PCT/CN2009/000830
Authority: WO
Inventors: 金崇庭; 刘峥嵘; 姜向东
Original assignee: 中兴通讯股份有限公司
Priority date: 2008-07-31
Filing date: 2009-07-24
Publication date: 2010-02-04
Also published as: EP2293644A1; EP2293644B1; CN101330340B; EP2293644A4; CN101330340A

Abstract

A method for adaptively adjusting receiving rate, buffering and playing of a mobile multimedia broadcast terminal includes the following steps: periodically detecting network state; calculating a receiving rate factor which indicates the level of the code rate of received audio and video data according to the network state; arranging a buffering queue; de-multiplexing received broadcasting modulation data to obtain the audio and video data; selectively decoding the audio and video data according to the receiving rate factor and storing the decoded audio and video data into the buffering queue; and playing the decoded audio and video data in the buffering queue. The mobile multimedia broadcast terminal automatically adjusts receiving rate according to the network state and arranges buffering queue to store buffering data, and thus buffering data in the buffering queue can be played in events such as signal is interrupted momentarily.

Description

Method for adaptively adjusting receiving rate buffer playing by mobile multimedia broadcasting terminal

The present invention relates to a mobile multimedia broadcast technology, and in particular to a method for adaptively adjusting a receive rate buffer play by a mobile multimedia broadcast terminal.

Background technique

With the continuous development of mobile communication and digital TV services, the mobile TV service is considered to be one of the "killer" applications in the third generation of mobile communication (3G) era, and has attracted much attention from the industry. The mobile TV service can become a converged service provided by the mobile communication industry and the broadcasting and television industry. Mobile operators can maximize the use of network capabilities and user management capabilities to provide richer services; broadcasters can take full advantage of content resources, expand user reach, and enable the mobilization of broadcast networks. At present, China's mobile phone users are close to 400 million, and the number of households with TV sets has reached 350 million. Faced with a large mobile user base and broadcast TV user base, network operators and equipment manufacturers are paying close attention to mobile TV services, hoping to become a new growth point for mobile multimedia services.

The transmission of audio and video data in a wireless environment with interference, error code, and packet loss, and the instability and unreliability of network services will affect the normal use and expansion of users. Existing mobile TVs based on broadcast TV network technology, due to incomplete coverage, unstable signals and weak transmission interference caused by bad weather, the unstable signal reception of mobile TV terminals will be inevitable, and the network has not been optimized. It's been there for a long time.

In addition, the existing mobile multimedia broadcast receiving terminal receives the broadcast television signal in real time, and if it encounters an unstable network, for example, an average of more than 3 times per minute for a signal reception failure of about 4 seconds, the downlink received signal strength indication (RSSI) ) Below the signal threshold, the program is prone to intermittent problems when playing. Moreover, when switching channels, the user must disconnect the original channel, connect and receive the new channel, and the TV of the original channel cannot be played during the period, and the user waits for a long time.

In summary, in the prior art, the defect of the quality of service (QoS) of the mobile multimedia broadcast network and the intermittent play of the program greatly affect the viewing experience of the mobile multimedia user, which is an important technical problem to be solved. Summary of the invention

The technical problem to be solved by the present invention is to provide a method for adaptively adjusting the reception rate buffer playback of a mobile multimedia broadcast terminal to improve the service quality of the mobile multimedia broadcast network.

In order to solve the above technical problem, the present invention provides a method for adaptively adjusting a receiving rate buffer playback of a mobile multimedia broadcast terminal, including:

Periodically detecting network status;

Calculating a reception rate factor indicating a code rate level of the received audio and video data according to the network state; setting a buffer queue;

Demultiplexing the received broadcast modulated data to obtain audio and video data;

Selecting and saving the audio and video data to the buffer queue according to the reception rate factor;

Playing the decoded audio and video data in the buffer queue.

In the method as described above, the step of calculating the reception rate factor according to the network state may include the terminal calculating the reception rate according to the received signal strength indication, the weak signal drop threshold, and the weak signal standby threshold. a factor, wherein the received signal strength indication is lower than the weak signal drop threshold, and a line is generated, and when the weak signal standby threshold is reached, it is in standby.

The method as described above may further comprise:

After the buffer queue is set, the capacity of the buffer queue is dynamically adjusted according to the receiving rate factor.

In the method as described above, the step of selectively decoding and saving the audio and video data into the buffer queue according to the reception rate factor may include pairing the audio and video according to the reception rate factor The data is hierarchically decoded and saved to the buffer queue.

Further, the step of hierarchically decoding the audio and video data may include decoding an intraframe compressed key frame in the video data, and partially or completely discarding the redundant slice data.

In the method as described above, the step of playing the decoded audio and video data in the buffer queue may include reading the audio and video data in the buffer queue and playing it in a first in first out manner.

In the method as described above, when the receiving rate factor changes, the The data queue is reconstructed by the rush queue to achieve playback synchronization.

The present invention also provides a method for a mobile multimedia broadcast terminal to switch channels, including: when the channel is switched, the terminal can disconnect the current channel, and perform the method for adaptively adjusting the receive rate buffer play of the mobile multimedia broadcast terminal, and play the Decoding the decoded audio and video data of the current channel in the buffer queue, after successfully connecting the target channel, performing the method for adaptively adjusting the receiving rate buffer playback of the mobile multimedia broadcast terminal, and playing the decoded sound of the target channel Video data.

Further, in the method of switching channels:

Receiving, by the terminal, a channel switching instruction;

When the terminal disconnects the current channel, playing the decoded audio and video data saved in the buffer queue by the current channel, and connecting the target channel;

After the connection to the target channel is successful, the decoded audio and video data of the current channel is stopped, and the decoded audio and video data of the target channel is played. BRIEF abstract

Figure 1 is a schematic illustration of the steps of the process of the invention.

2 is a schematic diagram showing the steps of an embodiment of the method of the present invention taking a CMMB terminal as an example.

FIG. 3 is a schematic diagram showing the steps of adaptively adjusting the reception rate factor according to the network state in the present invention. 4 is a schematic diagram showing the steps of an embodiment of channel switching in the present invention. Preferred embodiment of the invention

The embodiments of the present invention will be described in detail below with reference to the accompanying drawings and embodiments, by which the present invention can be applied to the technical solutions to the technical problems, and the implementation of the technical effects can be fully understood and implemented.

Figure 1 shows a schematic diagram of the steps of the method of the invention, comprising the steps of:

Step 110: periodically detect a network state, and calculate a receiving rate factor S _n indicating a code rate level of the received audio and video data according to the network state;

Step 120, setting a buffer queue FIFO and adjusting the buffer team according to the receiving rate factor S state The capacity of the column;

Step 130: Demultiplexing the received broadcast modulation data to obtain audio and video data, and selectively decoding and saving the audio and video data into a buffer queue FIFO according to the reception rate factor S _n ;

Step 140: Play the decoded audio and video data in the buffer queue FIFO.

The receiving rate factor S _n applied to the above steps reflects the network condition, indicating the received audio and video (AV) data rate level, and reflects the approximate duration that the buffered data in the buffer can be played. S _n in the range _{[0-S max], S} n value of 0 indicates the network signal is very good, the AV data without omissions, i.e. without buffering; ₈₁₁ larger value indicates worse condition of the network, the AV data The more the deletion, the longer the buffer delay and the stronger the anti-interruption capability. 8 Take S _{max to} indicate that the network signal is the worst, AV data is deleted the most, and the buffer delay is the longest. The calculation of s _n depends on signal strength or other factors that affect the continuity of video playback.

The adaptive rate adjustment method mainly performs partial decoding on the demultiplexed multimedia audio and video data according to the priority of the data frame according to the receiving rate factor s _n to obtain lower decoding complexity and lower spatial resolution. Lower time resolution (ie frame rate). For example, only the intraframe compressed I frame or the combined P frame in the video data is decoded, and the B frame and other redundant slice data are partially discarded or all discarded. When 0 < S _n <= S _max , partial decoding and different decoding are performed, and audio and video data of different code rates can be obtained and divided into S _n levels.

A FIFO buffer queue for storing multimedia data frames extracted by television data demultiplexing. The rate factor S _{n is} different, and the code rate for extracting multimedia data is also different. Therefore, a fixed-size FIFO buffer can accommodate a different number of data frames, thereby changing the buffer time. When the terminal plays, the first-in-first-out FIFO mode is used to read the buffer queue data for decoding and playing. When the rate factor S _n changes, due to the buffer time adjustment, it is necessary to indicate the playback synchronization, and the data frame reconstruction of the FIFO queue can achieve this purpose.

The following describes the method of the present invention by taking the China Mobile Multimedia Broadcasting Technology (CMMB) mobile TV terminal as an example. The CMMB terminal can operate in GSM/GPRS mobile communication network, CMMB ultra high frequency (UHF) band (470-862MHz) and S band (2635-2660MHz) CMMB broadcast TV mode. FIG. 2 shows a method for the CMMB terminal to adaptively adjust the receiving rate according to the network state to implement buffer playback. When the terminal is powered on and enters the mobile TV module, the user can set whether to enable or not. Network intermittent optimization (adaptive rate adjustment) function. If the user enables this function, the terminal starts collecting network data required for recording and judgment, adjusting the rate factor, selectively buffering demultiplexed data, decoding and playing, etc., and the steps are as follows:

Step 210: The terminal is powered on, and the mobile TV service is selected to connect to the program channel. Step 220: The terminal receives the broadcast modulation data, and demultiplexes the received broadcast modulation data to obtain audio and video data.

Step 230, determining whether to enable the buffer optimization function, if it is turned on, go to step 240, otherwise go to step 290;

Step 240, calculating the rate factor S _n according to the network state every T seconds;

Step 245, it is determined whether the rate factor 8 ₁₁ is greater than 0, if greater than 0, then go to step 250, otherwise (S _n is equal to 0) to step 270;

Step 250: Perform hierarchical decoding on the received data and store it in the buffer queue FIFO, and go to step 260 to execute;

Step 260, reading the audio and video data stored in the buffer queue FIFO and playing, and ending.

Step 270, S _n is equal to 0, indicating that the network condition is very good, determining whether data is stored in the buffer queue FIFO, and if yes, proceeding to step 280, otherwise proceeding to step 290;

Step 280, the data in the buffer queue is cleared, the subsequent play will no longer use the data therein, but the received demultiplexed data is directly used, and then the process proceeds to step 290;

In step 290, the audio and video data just received is directly decoded and played, and the process ends.

Further, step 240, if the calculated S _n is changed, then the FIFO buffer queue for re-divided, or divided into constant buffer.

Adaptive adjustment of the receive rate factor S 々 according to the network status, see Figure 3, below for example RSSI:

Step 310, acquiring a broadcast station downlink RSSI value (S _n initial value is 0, i.e., without buffering); step 320, it is determined whether RSSI is less than the threshold value dropped weak signal R _w, then the transfer step is 330, otherwise turn to Step 360, Wherein the weak signal drop threshold R _w indicates that the RSSI is lower than the R _w ; Step 330, further determining whether the RSSI is less than the continuous cumulative number of R _w is greater than the stability coefficient NS, then go to step 340, otherwise go to step 350;

Step 340, S _n takes a value of S _max , and returns to step 310 to continue execution;

Step 350, performing a self-addition operation on the continuous cumulative number of times, and returning to step 310 to continue; step 360, determining whether the RSSI is greater than the weak signal standby threshold R _hlgh , if _yes , _proceeding to step 370, otherwise _proceeding to step 399, wherein the weak signal standby valve The value R _hl gh indicates that the RSSI can be in the standby state when it reaches R _hlgh ;

Step 370, further determines whether RSSI is greater than the cumulative number _R hlgh continuous stability factor is greater than T _h NS, then go to step 380 is, otherwise, go to step 390;

Step 380, S _n takes a value of 0, and returns to step 310 to continue execution;

Step 390, the cumulative number of T _h continuous self-add operation returns to step 310 to continue; and

In step 399, the RSSI is between [R _w , R _hl gh], according to Ri. _w , R _hlg h and RSSI calculate S _n , one of the calculation methods is [ ( Rhigh

) / ( RSSI-R _w ) ] , then round the result.

If S _n 1 every T seconds, then the network is stable in the T * NS seconds, then there is no need to adjust the value S _n, otherwise readjust reception rate.

In a specific application example of the present invention, S _{max is} generally taken within 5 seconds to ensure that the TV program delay is within a range that the user can tolerate, and the amount of buffered data does not exceed the buffer space available by the terminal. T may be a value of 180 seconds, 300 seconds, 600 seconds, and the NS value of 3, then the corresponding S _n 540 seconds, 900 seconds, stable over 1800 seconds.

The invention is applied in the channel switching process, during the disconnection of the current channel to connect the new channel, the audio and video data saved in the buffer queue of the current channel can be played while switching, so that the broadcast program of the current channel is continuously played during the switching process, After switching to the new channel, the broadcast program of the new channel is played, thereby achieving the effect of reducing the user's switching waiting time. Figure 4 illustrates an embodiment of a channel switch that includes the following steps:

Step 410, when the terminal plays the current channel A, receives a channel switching instruction sent by the user, and wants to switch from the current channel A to the target channel B; Step 420, it is determined whether to enable the buffer optimization function, if it is turned on, then go to step 430, if not, then go to step 440;

Step 430, while disconnecting the current channel A, playing the audio and video data of the current channel A saved in the buffer queue, connecting to the target channel B, and then proceeding to step 450;

Step 440, disconnect the current channel A, and do not play the audio and video data in the buffer queue, connect the target channel B, and then go to step 450;

Step 450: After the target channel B is successfully connected, the program content of the target channel B is played. While the terminal is disconnecting the original channel, the terminal can continue to play the cached data of the original channel without stopping the playback immediately. Therefore, the present invention can smoothly switch between channel switching, reduce user waiting time, and greatly improve the user experience of mobile TV.

When the mobile multimedia receiving terminal is unstable in the network environment, the terminal side dynamically adjusts the receiving rate according to the network state, and saves the data in a buffer queue, and plays when the signal is intermittent or when the channel is switched. Buffering the cached data in the queue to avoid problems such as signal discontinuity, transmission delay, delay jitter, packet loss, etc. due to network signal instability or channel switching, improving QoS and providing users with Smooth audio and video playback, smooth and seamless channel switching, high-quality mobile multimedia end-user experience, and no need to retrofit existing networks, saving equipment costs.

While the embodiments of the present invention have been described above, the described embodiments are merely illustrative of the embodiments of the invention and are not intended to limit the invention. Any modification or variation in the form and details of the embodiments may be made by those skilled in the art without departing from the spirit and scope of the invention. The scope defined by the appended claims shall prevail.

Industrial applicability

Compared with the prior art, the method of the present invention automatically adjusts the receiving code rate according to the network state in the mobile multimedia broadcasting terminal, and sets the buffering queue to store the buffered data, and plays the buffer in the buffering queue in the case of a short intermittent signal. data.

Claims

Claim

A method for adaptively adjusting a receive rate buffer play by a mobile multimedia broadcast terminal, comprising: periodically detecting a network state;

Playing the decoded audio and video data in the buffer queue.

2. The method of claim 1 wherein:

The step of calculating a reception rate factor according to the network state, comprising: the terminal calculating the reception rate factor according to a received signal strength indication, a weak signal drop threshold, and a weak signal standby threshold, wherein the received signal strength A line is generated when the indication is lower than the weak signal drop threshold, and is in standby when the weak signal standby threshold is reached.

3. The method of claim 1 further comprising:

4. The method of claim 1 wherein:

The step of selectively decoding and saving the audio and video data into the buffer queue according to the reception rate factor includes hierarchically decoding and saving the audio and video data according to the reception rate factor In the buffer queue.

5. The method of claim 4, wherein:

The step of hierarchically decoding the audio and video data includes decoding the intraframe compressed key frame in the video data, and partially or completely discarding the redundant slice data.

6. The method of claim 1 wherein: The step of playing the decoded audio and video data in the buffer queue includes reading the audio and video data in the buffer queue and playing it in a first in first out manner.

7. The method of claim 1 wherein:

When the receiving rate factor changes, the data frame is reconstructed by the buffer queue to achieve play synchronization.

8. A method for a mobile multimedia broadcast terminal to switch channels, comprising:

The terminal disconnects the current channel when the channel is switched, and performs the method of claim 1, playing the decoded audio and video data of the current channel in the buffer queue, and after successfully connecting the target channel, executing claim 1 The method of playing the decoded audio and video data of the target channel.

9. The method of claim 8 wherein:

Receiving, by the terminal, a channel switching instruction;

After the connection to the target channel is successful, the decoded audio and video data of the current channel is stopped, and the decoded audio and video data of the target channel is played.